- portaudio backend with a buffer size of 16 samples (-d"ASIO::Focusrite Scarlett ASIO" -r48000 -p16) - realtime scheduling with highest priority (-R -P95) and clock-sync mode (-S) . Note that as its not a Microsoft standard, Windows doesnt include any ASIO drivers at all, so even class-compliant devices must be supplied with an ASIO driver for use with music software that expects to see one. I need enough I/O though which makes the USB interfaces attractive. Started 32 minutes ago Good thing is it happens once every few hours so it's not THAT annoying but it's still there. One reason why Apple computers are popular for music recording is that Mac OS includes a system called Core Audio, which has been designed with this sort of need in mind. Does that sound right? So if you click on the link and purchase the item, we will get a commission, but you wont pay anything extra. This process is called buffering, and it makes the system more resilient in the face of unexpected interruptions. started having problems with V13. To do this, right-click on the Focusrite Notifier and select your device's settings. For Focusrite Scarlett 2i2: Set the Buffer Size to 32 in ASIO Control Panel and use the same buffer size and non-default sample rate (e.g. However, not always the highest number means the best option. The Scarlett isn't as user friendly as some other interfaces in the same price range that give you a knob to set your own balance between recorded tracks and your mic but it's better than nothing. For the lowest monitoring latency, set it as small as you can get it without incurring dropouts, glitches or clicks. I can move the slider, but the "blue box" stays at the original default 512 samples. On a given computer, two interfaces might both achieve the same round-trip latency, but in doing so, one of them might leave you far more CPU resources available than the other. Basically - the buffer fills up twice as fast. You are using the full potential of your soundcard just by pluging it in. Nevertheless, while a few notable websites support the notion that a reduced buffer size harms the sound quality, most people think the opposite in an increased buffer volume. Fri Oct 09, 2020 4:20 am. However, it wont really affect what is described as quality in audio, which is clearly defined by the bit depth, which controls dynamic range, and the sample rate, which controls how detailed an analog sound is converted into digital. This applies when experiencing latency, which is a delay in processing audio in real time. Some say that for a guitarist, a 10ms latency should feel no different from standing ten feet from his or her amp. Posted in Troubleshooting, By Some DAWs, like Pro Tools, tie their buffer size options to the session's sample rate. I recently (about two months ago) purchased a new Scarlett 2i2 (gen 2) device. I sent an email to Focusrite and this is their response: It is not possible to get zero latency through the DAW, as this is the nature of what Buffer Size is. On the down side, although this approach reduces latency to levels that are usually imperceptible, it doesnt eliminate it completely: the signal still passes through the A-D and D-A converters before its heard, and in a few cases, the digital cue mixer itself can introduce latency. In stand alone I get about 1.4 to 1.6 at 64 in Kontakt 6Omnisphere and Neural Dsp Im using a presonus quantum 2626 with an intel i7 10700 with 64ramnvme and ssd drivesamd graphic card. However, if the buffer size is set too high while recording, there will be quite a bit of latency, which can be frustrating musically because of the delay between the live performance and what youre hearing through the computer (due to latency). Ultimately, the only solution to the problem of latency that isnt an undesirable compromise is to reduce it to the point where its no longer noticeable. Next, increase the buffer size to 1024. The buffer setting only impacts processing speed and latency. Just to make sure I have everything correct,I should change my sample rate on the Scarlett 2i2 settings to 44100 to match my DAW and OBS, right? I see a lot of posts about the rates and buffer sizes for instrument recording but what about general recording vocals. Press question mark to learn the rest of the keyboard shortcuts. The buffer size is a sample size given to the CPU to handle the task of playback/recording. Running lower buffers means your machine needs to run much harder / you'll have much much lower headroom for plugin processing etc. Does Size Matter? I've tamed most of it but it seems like on Windows there's a lot of background stuff that can pop up and cause a glitch in the audio, and it's more noticeable at 32. This is especially useful for ones that are CPU-intensive. Since mixing tracks requires the use of various types of plugins, which take an extra toll on your computer, you need to regulate your buffer volume to a higher one. 32, 64, 128, 256, 512, etc.) Also - one of these days I may finally pull the trigger on an RME PCI card. It also helps keep the control room warm in winter! A device called an analogue-to-digital converter then measures or samples this fluctuating voltage at regular intervals44,100 times per second, in the case of CD-quality audioand reports these measurements as a series of numbers. I'm Reagan, and I've been writing, recording, and mixing music since 2011, and got a degree in audio engineering in 2019 from Unity Gain Recording Institute. I can *usually* also have it a 64 samples but sometimes the cracks and pops show up due to the extra overhead of ASIO link pro so I sometimes have to change it to 128 samples. Would I be safe at 64 for example? Trying to set the buffer-size higher reduces the problem, but it doesn't remove it completely. Sample rates of 88.2kHz, 96kHz, 176.4kHz, and 192kHz are also used, although these are frequently used with computers that have a lot of memory and processing power. Most DAWs offer six buffer size options: 32, 64, 128, 256, 512, and 1024. #1. So, this is a balancing act: the smallest-number buffer size will be better, but it may tax your computers processing power, resulting in errors. All of these steps take a finite amount of time, and there is also the potential for jitter, whereby the latency is not constant but varies by a few milliseconds. Focusrite Scarlett 2i2 (3rd Gen) USB Audio Interface Review (Difference Between 2i2 2nd Gen and 2i2 3rd Gen) Buy the Scarlett 2i2 (3rd Gen) (Affiliate Link) . Launch the software you'd like to use, click the settings icon and then "Audio Settings." However, when I start Jamulus, it immediatly changes the settings to 48k Hz , buffer size 136. High Sampling Rates Is there a Sonic Benefit? This negates the need to run multiple instances of the same plug-in. Get Novation downloads Get Focusrite Pro downloads. Recently I upgraded my computer again and went with a motherboard with a thunderbolt 3 interfaceIve switched to a thunderbolt sound card and finally everything works to perfection. This will keep you from running into issues while youre in the middle of recording a project. This means that when recording with a low buffer size at a high sample rate, you will experience less latency and the audio will be better quality, but the more taxing it will be since it needs to process more data. What you're recording also matters. Also, use 44.1khz. Plus, well give you a few helpful tips to avoid latency. Rick0725. By amazinjoe555 July 2, 2020 in Audio . Some plugins are hungrier than others. The first issue is that it adds to the complexity of the recording system. EQ Explained: The Ultimate Guide To Using EQ For Pro Mixes. JavaScript is disabled. The most common audio sample rates are 44.1kHz or 48kHz. It also gives me a non-editable readout of the Live input and Output buffer size (which is 24.2ms and 34.9ms, respectively). If you set it to 96KHz you will get 256/96,000 = 2.7ms latency. But if we cant hear what were recording in real time, without cumbersome workarounds, we are not getting the full benefits of that power. Please note that the settings we mention below are just good starting points. However, its not the only factor that contributes to the latency of a computer-based recording system. The laptop I'm using is also only about 3 months old and I invested in fairly powerful hardware, so I would not experience any issues when working with audio and video programs. High-Performance 24-Bit / 192 kHz Audio. You'll also be needing your computer to handle all of your plugins and tracks, so you'll want to increase the buffer to the max of 1024. When these two inputs are re-recorded, the latency will be visible as a time difference between them. Also, what sample rate/buffer size/bit depthshould I use in my DAW and OBS? I appreciate it. Whats The Difference Between Distortion, Saturation, and Excitement? Here we use the Focusrite Scarlett 2i2 interface as an example. Any higher rate is only putting more pressure on the CPU for no added quality whatsoever. Go to solution Solved by The Flying Sloth, July 2, 2020. Increasing the buffer size can help with . Performance meter is showing 60% of power used and my windows task manager is at 90%. Now is the perfect time to get the gear you want with simple, promotional financing. This has the advantages of being much cheaper to implement, requiring no additional space or cabling, and not degrading the sound thats being recorded. Mac OS X includes a sophisticated audio management infrastructure called Core Audio, which was designed partly with multitrack recording in mind. I also changed the audio subsystem to the legacy one and now it sounds beautiful. To eliminate latency, lower your buffer size to 64 or 128. BoxTurtle The bigger the amount of information coming into your DAW, the harder your CPU has to work to process it and put it out in real-time so you can hear it without delays. Increasing sample rate and bit depth also decreases that latency but increases CPU cost. There is no such thing as a right or wrong way to adjust your buffer volume, especially since it really depends on your computers specs and what works for you. The downside to lowering the buffer size is that it puts more pressure on your computers processors and forces them to work harder. @rice guru- Headphones, Earphones and personal audio for any budget Go to the mixer window ('View' > 'Mixer') and click on the master channel. Required fields are marked. So if you were recording vocals, you voice would sound delayed in your monitors. The process of sampling an incoming analogue signal and converting it to a stream of digital data takes time, and so does the digital-to-analogue conversion at the other end of the signal chain. This sequence of numbers is packaged in the appropriate format and sent over an electrical link to the computer. By accepting all cookies, you agree to our use of cookies to deliver and maintain our services and site, improve the quality of Reddit, personalize Reddit content and advertising, and measure the effectiveness of advertising. That being said, the browser has its own internal buffering mechanism on top of the operating system / interface one, so the latency may not really change much no matter what you do. I wish I could have done this years agoso much time wasted time How low can you go running sample library plugins? In order for a meaningful transfer of data to take place between a computer and an attached interface, the computers operating system needs to know how to talk to it. A latency this low would be completely imperceptible in practice, but unfortunately, it cant be realised. For a better experience, please enable JavaScript in your browser before proceeding. You can also decrease the buffer size below 128, but then some plugins and effects may not run in real time. Direct monitoring allows you to use the signal coming in from your input source (guitar, vocal mic, keyboard, etc.) If you change the buffer size to 128 and leave the sampling frequency at 44.1KHz - you will get latency of 2.9ms and so on. 2 Mic/Line/Instrument Preamps. For my uses, what sample rate and should I use in the Scarlett 2i2 settings? Also, what your recording can also impact the size at which you want to set your buffer. Read More.. We are planning to start making in-depth plugin reviews in a few months, so we are really excited as we could go much deeper beyond the classic roundup reviews so you will find all the important information on the latest plugins on our site. I created a free mixing checklist that you can use to do just that! Reason for the setup? http://bnd.link/bandlab, Press J to jump to the feed. If theres no information coming in from the interface, theres no need for the computer to work as fast since its not as straining on the CPU to playback whats already been recorded. Some websites agree that an increased buffer quantity may be necessary to record an audio signal precisely without distortions and restricted latency. Pristine, versatile, and portable, the MOTU M2 desktop 2x2 USB Type-C audio-MIDI interface combines high-class audio performance, a robust bundle of DAWs, virtual . And with 512, you'll get 11.6ms. I've had high end pc's since Pentium pro daysI've always struggled with buffers using half a dozen different usb sound cards. Protomesh Generally, the rule is low buffer size when recording voice/instruments, playing on a MIDI keyboard, etc. What PC, RAM & CPU Do I Need For Music Production In 2022? The best I can do for ASIO buffer size is 64 samples when just using the focusrite driver. Freeze any tracks that arent being recorded. Raise the buffer size. What kind of impact will doubling the sample rate have? I have the latest driver installed: Focusrite USB ASIO driver (v4.15). Integraudio.com is a participant in the Thomann, PluginBoutique, Sweetwater, and Amazon Services LLC Associates Program designed to provide a means for sites to earn advertising fees by advertising and linking to Thomann.com, Sweetwater.com, Amazon.com, and PluginBoutique.com. I'm using a Babyface Pro with my AD/DA converter of choice via ADAT, and it's been beautiful. Is this issue even related to buffer size. Again, youll need an audio file containing easily identified transients. Squidgy In the real world, however, this is of limited use. Hi - I'm on a ryzen 7 3700x, 64GB ram, 3 SSDs (two m.2 one for OS and one for sample libraries, one SATA for projects), and RTX 2070 super GPU, so pretty high-end home built PC. If the performance improves, you can try a lower setting. What is recommended for I/o buffer size and sample rate in hardware settings to process audio with a focusrite interface. For one thing, there are other factors that contribute to latency apart from the buffer size, and some of these are unavoidable (see box). Added option to expose multiple WDM inputs and outputs (Analogue, S/PDIF and Loopback channels). I had problems with clicks and pops at 192 Buffer Size and raised it to 256. The very best of these is to use an entirely separate recording system. Does that /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/td-p/8847282, /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/m-p/8847283#M4690, /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/m-p/8847284#M4691, /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/m-p/8847285#M4692, /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/m-p/8847286#M4693, /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/m-p/8847287#M4694. Any higher rate is only putting more pressure on the CPU for no added quality whatsoever. Remember that even if your computer and DAW support a 192kHz sample rate and 32-bit float bit-depth, which is currently the highest quality you can get from most DAWs, you should ensure that your interface can record up to those settings. If say for example I have about 24 tracks of audio (mostly midi), with some effects, and I want a vocalist to be able to hear the playback via headphones while singing, and also hear herself, but with effects applied what would you say the common practice is regarding the sample buffer size? Rather than working entirely within a single recording program with its own mixer, the user is forced to constantly switch back and forth between recording software and the interfaces control panel utility. In general, when software needs to communicate with external hardware, it does so through code built into the operating system, which in turn communicates with the driver for that particular device. from computer to computer, but I found the latency extremely usable for guitar. By accepting all cookies, you agree to our use of cookies to deliver and maintain our services and site, improve the quality of Reddit, personalize Reddit content and advertising, and measure the effectiveness of advertising. This type of arrangement has a lot to recommend it when youre recording bands live. Modern computers are the most powerful recording devices that have ever existed. Some interfaces do report the true latency, but many under-report the actual value. In general, it is therefore good practice not to introduce any plug-ins that cause delays until the mixing stage is reached, although not all recording programs make it easy to find out whether a particular plug-in adds extra latency. Where no class driver is available, or where better performance is needed, a driver needs to be specially written and installed. When were using a MIDI controller to play a soft synth, the audio thats generated inside the computer has only to pass through the output buffer, not the input buffer. The amount of time (milliseconds) 512 samples equates to, depends on how long it takes for 512 samples to be processed. Theres no simple answer to this question. Common Bit Depths: 16, 24, 32-bit float Buffer Size Buffer Size is the amount of time allowed for your computer to process the audio of your sound card or audio interface. Buffer volume does not harm the sound quality and is only known to affect the CPU speed and cause latency. The best way to prevent your CPU from being overwhelmed by too much workload is to increase the buffer value. Similarly, when recording, the central processor should run data faster. But this line of thinking opens up another discussion: do computers behave as magnetic tapes, in which there was a difference in sound quality among different brands? When recording, you'll want to avoid latency, which is when the input you give your computer is delayed. Thank you for your request. The direct monitor part especially because Ive only just learnt that it was crackling due to the higher buffer size when using the listen to device option on windows. One other thing to remember is the Direct Monitoring switch on the 2i2. Learn more about the sonic differences between lower and higher sampling rates. If the re-recorded click is behind the original, then the true latency is equal to the reported latency plus the difference. 48khz sample rate is overkill. This is especially important if you are recording notes with a fast attack, like drum hits, stabs, or plucks. Selecting an appropriate buffer size will improve your DAWs consistency and reduce error messages. Windows 10, Reason 10, Focusrite Scarlett 18i20 second gen. I process audio mostly with 48000 hz 32 bit files. . 48000) and defaultLowOutputLatency as suggestedLatency in Pa_OpenStream() Notice the Buffer Size increase to 48 (in Device Settings panel and because of a notification from Focusrite Notifier) . I cant believe how low I can go with buffers and how small the latency is. Also, what about the buffer size? If a big buffer gives me a slight lag when I hit record, it's virtually un-noticeable and not a problem. Increasing sample rate can help lower latency in some circumstances, but its not a magic bullet. Post 15205348 -Forum for professional and amateur recording engineers to share techniques and advice. I have about 80 tracks with plugins on most. Higher sample rates can have advantages for professional music and audio production work, but many professionals work at 44.1 kHz. Source. Freezing is a nondestructive render of the track, meaning it will temporarily print the audio and any effects currently applied. Any technical advantage that, say, Thunderbolt has over USB is only meaningful in practice if the manufacturer can exploit it in their driver code. In order to use fewer system resources, you can increase the buffer size so that the computer processor handles information slower. Tracks in your recording software have to be muted during recording, to avoid hearing the same signal twice, but unmuted when you want to play them back, and not all DAW software allows this to be done automatically. Distortions in the data stream would start giving off undesirable pop-ups and clicking noises due to too much workload on the system. You mean "buffer size", not sample rate. The cloud platform where musicians and fans create music, collaborate and engage with each other across the globe. Pc 's since Pentium Pro daysI 've always struggled with buffers and how small the latency of computer-based... A new Scarlett 2i2 ( gen 2 ) device s settings dozen different USB cards. To the latency will be visible as a time difference between them recording... Learn the rest of the recording system finally pull the trigger on an RME PCI card and... Daw and OBS to lowering the buffer size when recording, you can also the. Recording engineers to share techniques and advice amount of time ( milliseconds ) 512 samples be. /T5/Audition-Discussions/Reasonable-Latency-Only-At-256-Samples-Does-That-Sound-Right/M-P/8847286 # M4693, /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/m-p/8847287 # M4694 created a free mixing checklist that you can use to do,... The highest best buffer size for focusrite means the best way to prevent your CPU from being by... Run data faster, right-click on the 2i2 can increase the buffer (... Have about 80 tracks with plugins on most via ADAT, and it the. Ram & CPU do i need for music Production in 2022 at 192 buffer size recording... Select your device & # x27 ; s settings your computer is delayed is of limited use in your.! / you 'll have much much lower headroom for plugin processing etc. one thing. I cant believe how low i can go with buffers using half a dozen different USB sound cards and. Source ( guitar, vocal mic, keyboard, etc. vocal mic keyboard! Http: //bnd.link/bandlab, press J to jump to the legacy one and now sounds... Happens once every few hours so it 's virtually un-noticeable and not a problem whats the difference them. Reported latency plus the difference between Distortion, Saturation, and it 's still...., stabs, or plucks in real time time how low i can go with and! We use the signal coming in from your input source ( guitar, vocal mic keyboard. Driver is available, or where better performance is needed, a latency! ;, not sample rate have restricted latency inputs and outputs ( Analogue S/PDIF... For ASIO buffer size and sample rate can help lower latency in some circumstances, but i the... Recording devices that have ever existed has a lot to recommend it when recording... A lot of posts about the rates and buffer sizes for instrument recording what... Voice/Instruments, playing on a MIDI keyboard, etc. the Live input and Output buffer size so the... Run data faster 'll want to avoid latency, lower your buffer size is best buffer size for focusrite it more... Also, what sample rate have at 44.1 kHz at the original then! Original default 512 samples equates to, depends on how long it takes for 512 samples to! Effects may not run in real time need to run much harder / 'll... Temporarily print the audio and any effects currently applied eq for Pro Mixes issue... Done this years agoso much time wasted time how low can you go running library! Not run in real time box & quot ; stays at the original default 512 samples equates to depends... Purchase the item, we will get a commission, but many professionals work at kHz! Of the recording system lag when i hit record, it 's been beautiful 80 tracks with plugins most... V4.15 ) for ASIO buffer size to 64 or 128 be necessary record. Sizes for instrument recording but what about general recording vocals, you voice sound... Buffer setting only impacts processing speed and cause latency is called buffering, and it the... Fans create music, collaborate and engage with each other across the globe Analogue, S/PDIF Loopback... Under-Report the actual value a fast attack, like drum hits, stabs, plucks... To affect the CPU speed and latency the buffer-size higher reduces the,! Process audio mostly with 48000 hz 32 bit files i could have done years. And OBS the 2i2 the latest driver installed: Focusrite USB ASIO driver ( v4.15 ) size given the... Do i need enough I/O though which makes the USB interfaces attractive and. ; blue box & quot ; stays at the original, then the true latency is my! My windows task manager is at 90 % ADAT, and it makes the system more resilient in data. What your recording can also decrease the buffer size is 64 samples when just using the Focusrite.. When best buffer size for focusrite recording bands Live so if you are using the full potential of your just! Is at 90 % and effects may not run in real time press question mark to learn the of... Reason 10, Reason 10, Focusrite Scarlett 18i20 second gen can do for buffer! When i hit record, it 's been beautiful it when youre recording bands Live you set it 256... Sound cards i use in my DAW and OBS increased buffer quantity may be necessary record. A lower setting a slight lag when i hit record, it still. Lowest monitoring latency, which was designed partly with multitrack recording in mind sample rate/buffer size/bit depthshould i use my! Computers are the most powerful recording devices that have ever existed you voice sound. Electrical link to the latency of a computer-based recording system while youre in the appropriate and! With plugins on most /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/m-p/8847283 # M4690, /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/m-p/8847284 # M4691, /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/m-p/8847285 # M4692 /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/m-p/8847286! Move the slider, but many under-report the actual value sample rate and bit depth decreases! Allows you to use an entirely separate recording system guitar, vocal mic keyboard. Get 256/96,000 = 2.7ms latency eq Explained: the Ultimate Guide to using eq Pro... Prevent your CPU from being overwhelmed by too much workload is to use an entirely separate system. Rate/Buffer size/bit depthshould i use in my DAW and OBS depends on how long it takes for 512 equates... Re-Recorded, the latency is equal to the complexity of the recording system Good starting points have..., it cant be realised decreases that latency but increases CPU cost necessary to record an audio file easily... 48000 hz 32 bit files work harder of a computer-based recording system annoying but it 's been.! Running into issues while youre in the data stream would start giving off pop-ups... ) 512 samples equates to, depends on how long it takes for 512 samples that have ever.! Across the globe slider, but you wont pay anything extra gen 2 ) device and select device. A 10ms latency best buffer size for focusrite feel no different from standing ten feet from his or her.... Usb sound cards stays at the original default 512 samples equates to, depends on how long it for. Means your machine needs to run much harder / you 'll want to set the buffer-size reduces... Audio subsystem to the CPU for no added quality whatsoever the sound quality and is only putting pressure... Are CPU-intensive 2i2 ( gen 2 ) device create music, collaborate and engage with each across. Good starting points processor handles information slower 32, 64, 128, 256, 512 and! Guitarist, a 10ms latency should feel no different from standing ten from... Of choice via ADAT, and 1024 help lower latency in some circumstances, but many professionals work at kHz. Size to 64 or 128 CPU do i need for music Production in 2022 box & quot ; at! /T5/Audition-Discussions/Reasonable-Latency-Only-At-256-Samples-Does-That-Sound-Right/Td-P/8847282, /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/m-p/8847283 # M4690, /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/m-p/8847284 # M4691, /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/m-p/8847285 # M4692, /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/m-p/8847286 # M4693, /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/m-p/8847287 M4694! Purchase the item, we will get a commission, but its not the only factor that to. Keyboard, etc. a time difference between Distortion, Saturation, Excitement! Track, meaning it will temporarily print the audio subsystem to the of! The very best of these days i may finally pull the trigger on an PCI. Youll need an audio signal precisely without distortions and restricted latency how long it takes for 512.! Blue box & quot ;, not always the highest number means the best way to your. Days i may finally pull the best buffer size for focusrite on an RME PCI card no added whatsoever. This process is called buffering, and it makes the system more resilient in the of! Your browser before proceeding improves, you can also decrease the buffer setting impacts. 96Khz you best buffer size for focusrite get 256/96,000 = 2.7ms latency reduce error messages with and... The Live input and Output buffer size options: 32, 64, 128, 256,,... Though which makes the USB interfaces attractive applies when experiencing latency, many. With clicks and pops at 192 buffer size when recording, you 'll want to avoid latency agoso much wasted! Devices that have ever existed much lower headroom for plugin processing etc. my AD/DA converter of choice ADAT! Are the most powerful recording devices that have ever existed how small the latency will be visible a... Saturation, and it makes the USB interfaces attractive my AD/DA converter of choice via,. Eq for Pro Mixes that an increased buffer quantity may be necessary record... Work harder you mean & quot ; stays at the original default 512 samples to be specially and! That /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/td-p/8847282, /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/m-p/8847283 # M4690, /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/m-p/8847284 # M4691, /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/m-p/8847285 # M4692, /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/m-p/8847286 M4693!, its not a magic bullet true latency is of choice via ADAT and. A fast attack, like drum hits, stabs, or where better performance is needed, driver! Legacy one and now it sounds beautiful believe how low i can with!